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Author	SHA1	Message	Date
Walter Doekes	5bf283d406	Clean up Makefile "warning" clutter when makeopts doesn't exist. ... Review: https://reviewboard.asterisk.org/r/2304 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a5df2542c3	Don't attempt a websocket protocol removal if res_http_websocket isn't there ... This patch sets the protocols container provided by res_http_websocket to NULL when the module gets unloaded and adds the necessary checks when adding/ removing a websocket protocol. This prevents some FRACKing on an invalid pointer to the disposed container if a module that uses res_http_websocket is unloaded after it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b8d4e573f1	Add multi-channel Stasis messages; refactor Dial AMI events to Stasis ... This patch does the following: * A new Stasis payload has been defined for multi-channel messages. This payload can store multiple ast_channel_snapshot objects along with a single JSON blob. The payload object itself is opaque; the snapshots are stored in a container keyed by roles. APIs have been provided to query for and retrieve the snapshots from the payload object. * The Dial AMI events have been refactored onto Stasis. This includes dial messages in app_dial, as well as the core dialing framework. The AMI events have been modified to send out a DialBegin/DialEnd events, as opposed to the subevent type that was previously used. * Stasis messages, types, and other objects related to channels have been placed in their own file, stasis_channels. Unit tests for some of these objects/messages have also been written. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a2a53cc306	Stasis application WebSocket support ... This is the API that binds the Stasis dialplan application to external Stasis applications. It also adds the beginnings of WebSocket application support. This module registers a dialplan function named Stasis, which is used to put a channel into the named Stasis app. As a channel enters and leaves the Stasis diaplan application, the Stasis app receives a 'stasis-start' and 'stasis-end' events. Stasis apps register themselves using the stasis_app_register and stasis_app_unregister functions. Messages are sent to an application using stasis_app_send. Finally, Stasis apps control channels through the use of the stasis_app_control object, and the family of stasis_app_control_* functions. Other changes along for the ride are: * An ast_frame_dtor function that's RAII_VAR safe * Some common JSON encoders for name/number, timeval, and context/extension/priority Review: https://reviewboard.asterisk.org/r/2361/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	426095bc55	Add a res_sorcery_astdb module which uses the astdb to persist objects. ... Review: https://reviewboard.asterisk.org/r/2420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	03286cf23f	Fix For Not Overriding The Default Settings In chan_sip ... The initial report was that the "nat" setting in the [general] section was not having any effect in overriding the default setting. Upon confirming that this was happening and looking into what was causing this, it was discovered that other default settings would not be overriden as well. This patch works similar to what occurs in build_peer(). We create a temporary ast_flags structure and using a mask, we override the default settings with whatever is set in the [general] section. In the bug report, the reporter who helped to test this patch noted that the directmedia settings were being overriden properly as well as the nat settings. This issue is also present in Asterisk 1.8 and a separate patch will be applied to it. (issue ASTERISK-21225) Reported by: Alexandre Vezina Tested by: Alexandre Vezina, Michael L. Young Patches: asterisk-21225-handle-options-default-prob_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2385/ ........ Merged revisions 384827 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b8e5189456	Separate some event struct definitions from instantiation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6a25d49296	chan_dahdi: Change inband_on_proceeding option default to no/disabled. ... (issue ASTERISK-21151) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	79818112fd	chan_dahdi: Add inband_on_proceeding compatibility option. ... The new inband_on_proceeding option causes Asterisk to assume inband audio may be present when a PROCEEDING message is received. Q.931 Section 5.1.2 says the network cannot assume that the CPE side has attached to the B channel at this time without explicitly sending the progress indicator ie informing the CPE side to attach to the B channel for audio. However, some non-compliant ISDN switches send a PROCEEDING without the progress indicator ie indicating inband audio is available and assume that the CPE device has connected the media path for listening to ringback and other messages. ASTERISK-17834 which causes this issue was dealing with a non-compliant network switch. (closes issue ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett ........ Merged revisions 384685 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384689 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1cafccdf87	Update documentation for CHANNEL function ... Document that you can read/write the 'accountcode' and 'amaflags' on a channel. ........ Merged revisions 384640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384641 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b9962ee26a	astobj2: Fix rbtree duplicate handling. ... OBJ_PARTIAL_KEY searching a rbtree did not find all possible matches if the container did not accept duplicates. Added matching node bias to indicate which matching node is being searched for: first, last, any. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	0da0797cb0	Fixed spurious rebuilds of func_version. ... func_version.so was being rebuilt every time, because build.h was changing every build, because of the cleantest dependency that was added in r384410 to fix parallel make bugs. Now build.h will only be created if it does not exist, which was the original behavior of the Makefile. ........ Merged revisions 384544 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384545 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	56313ee068	Pass the object type name to the configuration framework. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	bcc0aca23d	Make things work again ... Sorry folks. ',' are still greater than '|'. Thanks for playing along :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	d4e25a456e	install_prereq: Build jansson from source, when necessary ... When r383579 was committed, it made Jansson a required dependency. While libjansson-dev and jansson-devel are available on recent distros, some older (but still supported) distros don't have it. There's a pull request[1] to get it into repoforge, but that still doesn't help everyone. (And helps no one until the pull request is merged and packages are built). This patch adds Jansson install from source to the install_unpackaged() function. There are a few gotcha's, which makes this change not completely trivial. * Since Jansson may be installed by a package, don't install from source if a package installation can be found * libresample may also be installed via package, so I added a similar check to that. * Since Jansson installs into /usr/local, this patch also adds /usr/local/lib to /etc/ld.so.conf.d so that the library can be found. * The alternative was to install into /usr, but then it gets complicated having to deal with EL's /usr/lib{32,64} shenanigans. [1]: https://github.com/repoforge/rpms/pull/250 Review: https://reviewboard.asterisk.org/r/2414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8c5367226b	Make appropriate items parse using '|' instead of ',' ... This patch fixes a bug introduced in r76703, wherein Asterisk could only parse arguments in the so-called 'recommended' way, e.g., NoOp(foo,bar). The proper syntax of NoOp,foo|bar is now parsed correctly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3f2ff8594b	Remove silly use of strncmp. ... ........ Merged revisions 384414 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	b23e8e1950	stasis: Fixed message ordering issues when forwarding ... This patch fixes an issue of message ordering that occurs when multiple topics are forwarded to an aggregator topic (such as ast_channel_topic_all()). It is (very reasonably) expected that the rules governing message dispatch order still apply, so long as the messages start from the same thread, and are received by the same subscription. Because the existing code had an additional layer of dispatching via the Stasis thread pool for forwards, those promises couldn't be kept. Forwarding subscriptions no longer have their own mailbox, and now dispatch directly from the forwarding topic's stasis_publish() call. This means that the topic's lock is held for the duration of not only a message's dispatch, but the dispatch of all the forwards. This shouldn't be a problem right now, but if an aggregator topic had many subscribers, it could become a problem. But I figure we can write more clever code when the time comes, if necessary. Review: https://reviewboard.asterisk.org/r/2419/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2d45dbc79b	Fix parallel make problems. ... Occasionally, make -j would fail due to missing includes, or other unusual errors. This was due to the 'cleantest' target, which was designed to force a make clean when some change in the code would cause the typical depedency checking to fail. Several targets in the main Makefile did not depend upon cleantest, hence would run in parallel to it. By adding the dependency, make -j runs happily now. Review: https://reviewboard.asterisk.org/r/2418/ ........ Merged revisions 384410 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384411 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ad191ebfcd	Properly format an intmax_t value ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e8015cc460	Convert TestEvent AMI events over to Stasis Core ... This patch migrates the TestEvent AMI events to first be dispatched over the Stasis-Core message bus. This helps to preserve the ordering of the events with other events in the AMI system, such as the various channel related events. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d16efd5be8	app_voicemail: Add blank argument to externnotify if no context argument ... At least one call to run_externnotify provides a NULL context parameter and because the snprintf statement doesn't account for a NULL context parameter, it simply writes '(null)' to the arguments string instead. This patch makes it write two quotes back to back for that argument instead in the event of a NULL context. (closes issue ASTERISK-18207) Reported by: Barry L. Kline Patches: modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930) ........ Merged revisions 384325 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384326 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a1c94fece8	Add uuid wrapper API call ast_uuid_generate_str(). ... * Updated test_uuid.c to test the new API call. * Made system use the new API call to eliminate "10's of lines" where used. * Fixed untested ast_strdup() return in stasis_subscribe() by eliminating the need for it. struct stasis_subscription now contains the uniqueid[] string. * Fixed some issues in exchangecal_write_event(): Create uid with enough space for a UUID string to avoid a realloc. Fix off by one error if the calendar event provided a UUID string. There is no need to check for NULL before calling ast_free(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	71206544a7	Break the world. Stasis message type accessors should now all be named correctly. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1a2a4578d2	Convert MWI state message type to the new stasis naming convention ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c67a06a2ff	Added a doxygen group for Stasis messages and topics ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	72bccf69c3	Address uninitialized conditional that valgrind found ... ........ Merged revisions 384162 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384163 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	15b892323a	Fix a file descriptor leak in off nominal path ... While looking at the security vulnerability in ASTERISK-20967, Walter noticed a file descriptor leak and some other issues in off nominal code paths. This patch corrects them. Note that this patch is not related to the vulnerability in ASTERISK-20967, but the patch was placed on that issue. (closes issue ASTERISK-20967) Reported by: wdoekes patches: issueA20967_file_leak_and_unused_wkspace.patch uploaded by wdoekes (License 5674) ........ Merged revisions 384118 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384119 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	15bbfb941f	Fix white noise on SRTP decryption ... When res_rtp_asterisk.c was altered to avoid attempting to apply unprotect algorithms to non-audio RTP packets, the test used was incorrect. This caused the audio packets to not be decrypted and resulted in loud white noise on the other endpoint (or both endpoints depending on the call legs involved). The test now properly checks the version field in the RTP header to ensure that RTP and RTCP are decrypted while other types of packets are not. (closes issue ASTERISK-21323) Reported by: andrea Tested by: Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff uploaded by Kinsey Moore ........ Merged revisions 384048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384049 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	0ffce56f1b	AST-2013-003: Prevent username disclosure in SIP channel driver ... When authenticating a SIP request with alwaysauthreject enabled, allowguest disabled, and autocreatepeer disabled, Asterisk discloses whether a user exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The information is disclosed when: * A "407 Proxy Authentication Required" response is sent instead of a "401 Unauthorized" response * The presence or absence of additional tags occurs at the end of "403 Forbidden" (such as "(Bad Auth)") * A "401 Unauthorized" response is sent instead of "403 Forbidden" response after a retransmission * Retransmission are sent when a matching peer did not exist, but not when a matching peer did exist. This patch resolves these various vectors by ensuring that the responses sent in all scenarios is the same, regardless of the presence of a matching peer. This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of the testing and the solution to this problem was done by Walter as well - a huge thanks to his tireless efforts in finding all the ways in which this setting didn't work, providing automated tests, and working with Kinsey on getting this fixed. (closes issue ASTERISK-21013) Reported by: wdoekes Tested by: wdoekes, kmoore patches: AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674) ........ Merged revisions 384003 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ec144089ea	AST-2013-002: Prevent denial of service in HTTP server ... AST-2012-014, fixed in January of this year, contained a fix for Asterisk's HTTP server for a remotely-triggered crash. While the fix put in place fixed the possibility for the crash to be triggered, a denial of service vector still exists with that solution if an attacker sends one or more HTTP POST requests with very large Content-Length values. This patch resolves this by capping the Content-Length at 1024 bytes. Any attempt to send an HTTP POST with Content-Length greater than this cap will not result in any memory allocation. The POST will be responded to with an HTTP 413 "Request Entity Too Large" response. This issue was reported by Christoph Hebeisen of TELUS Security Labs (closes issue ASTERISK-20967) Reported by: Christoph Hebeisen patches: AST-2013-002-1.8.diff uploaded by mmichelson (License 5049) AST-2013-002-10.diff uploaded by mmichelson (License 5049) AST-2013-002-11.diff uploaded by mmichelson (License 5049) ........ Merged revisions 383978 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4b5a0e1932	AST-2013-001: Prevent buffer overflow through H.264 format negotiation ... The format attribute resource for H.264 video performs an unsafe read against a media attribute when parsing the SDP. The value passed in with the format attribute is not checked for its length when parsed into a fixed length buffer. This patch resolves the vulnerability by only reading as many characters from the SDP value as will fit into the buffer. (closes issue ASTERISK-20901) Reported by: Ulf Harnhammar patches: h264_overflow_security_patch.diff uploaded by jrose (License 6182) ........ Merged revisions 383973 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	63a4da4eba	Fix skinny encall button to not blind xfer. ... The softbutton endcall should not turn a transfer into a blind transfer but hangup the exten being called and leave the original call on hold. This does that. (closes issue ASTERISK-21321) Reported by: wedhorn Tested by: snuffy, myself Patches: skinny-xferendcall01.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7aab90b366	Remove the noop handler from sorcery so it does not produce an empty value. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	58ee2b7d11	Resolve deadlock between SIP registration and channel based functions ... In r373424, several reentrancy problems in chan_sip were addressed. As a result, the SIP channel driver is now properly locking the channel driver private information in certain operations that it wasn't previously. This exposed two latent problems either in register_verify or by functions called by register_verify. This includes: * Holding the private lock while calling sip_send_mwi_to_peer. This can create a new sip_pvt via sip_alloc, which will obtain the channel container lock. This is a locking inversion, as any channel related lock must be obtained prior to obtaining the SIP channel technology private lock. Note that this issue was already fixed in Asterisk 11. * Holding the private lock while calling sip_poke_peer. In the same vein as sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing the same locking inversion. Note that this locking inversion typically occured when CLI commands were run while a SIP REGISTER request was being processed, as many CLI commands (such as 'sip show channels', 'core show channels', etc.) have to obtain the channel container lock. (issue ASTERISK-21068) Reported by: Nicolas Bouliane (issue ASTERISK-20550) Reported by: David Brillert (issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue ASTERISK-21296) Reported by: Gabriel Birke ........ Merged revisions 383863 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383878 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ec7de8ed97	Resolve deadlock between pending CDR and batch CDR locks ... r375757 attempted to resolve a race condition between multiple submissions of CDRs while in batch mode from attempting to destroy the scheduled batch submission by extending the batch CDR lock. Unfortunately, this causes a deadlock between the pending CDR lock and the batch CDR lock. This patch resolves the intent of r375757 by simply providing a new lock that protects the scheduling of the batches. The original batch CDR lock is kept to protect manipulation of the batch CDR settings, but has been placed such that it is not held when the pending lock is held. Thanks to Chase Venters for providing lock analysis on the issue. (issue ASTERISK-21162) Reported by: Chase Venters ........ Merged revisions 383839 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383840 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	88874a95d7	Suppress compiler warning. ... This code caused a compiler warning when --enable-dev-mode was not used. The warning was that this variable was set but not used. That was indeed the case as the only place this is used is as an argument to SKINNY_DEBUG which is compiled out when not in dev mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	03047a47b6	Fix multi-station answer race condition. ... When an SLA trunk is ringing (inbound call on the trunk) Asterisk will make outbound calls to the stations that have that trunk. If more than one station answers the call at the same time, all channels other than the first one to answer are left in a bad state. The channel gets leaked, is not connected to anything, and there's no way to get rid of it. We now properly clean up these losing channels by hanging up on them. Since they lost the race, as we process their answer, there is no ringing trunk for them to answer. ........ Merged revisions 383835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383836 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	23f363fcb1	Set the CALLERID(dnid-num-plan) for incoming ISDN calls. ... The CALLEDTON channel variable is set for incoming ISDN calls to the lower 7 bits of the Q.931 type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan) should have the same value. (closes issue ASTERISK-21248) Reported by: rmudgett ........ Merged revisions 383796 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383798 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	f073c27b60	Fix typo ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383754 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4227863d9a	Fix missing ' ' around '=' ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383753 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c2ae4acb15	install_prereq: removed some out-of-date comments ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2e0f5cc854	install_prereq: Adding jansson-devel to RH packages ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4a6237b231	Move NewCallerid, HangupRequest and SoftHangupRequest to Stasis ... HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis messages, with the cause code as an optional field in the blob. NewCallerid now simply watches for changes in the callerid information in channel snapshots, and creates the AMI event appropriately. Since the original NewCallerid event honored the channelvars setting in manager.conf, the channel variables configured there had to become a part of the channel snapshot. These are now a part of every snapshot based event, making the configuration description "every time a channel-oriented event is emitted" less of a lie. There a a few other changes wrapped up in here as well. * When ast_channel_topic() is given NULL for a channel, it returns the ast_channel_topic_all() topic instead of NULL. This can clean up a lot of NULL checking we're doing currently. * The fields Cause and Cause-txt were removed from the base channel information and put only on the Hangup events, since those fields are meaningless outside of a Hangup event. * Removed the pipe-delimiter processing of the channelvars field, since that's been deprecated forever. (closes issue ASTERISK-21096) Review: https://reviewboard.asterisk.org/r/2405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	d484f366f5	Properly delimit post data in res_config_curl. ... ........ Merged revisions 383667 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383668 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383669 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	766c146fe3	Fixed another issue from r383579. ... Core modules don't honor <depend> flags in MODULEINFO, which broke jansson if specified --with-jansson to configure. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	2a65c9408c	Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On A Channel ... A regression was accidentally introduced when allowing an optional ID to be used when calling StopMixMonitor. When we are unable to stop MixMonitor on a channel, -1 is being returned which triggers the hangup of the channel. This patch restores the prior behavior by returning 0 whether we were successful or not. It also allows the call from the manager to use the return code when the action fails. (closes issue ASTERISK-21294) Reported by: daroz Tested by: daroz Patches: asterisk-21294-stop_mixmonitor_hangingup.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2404/ ........ Merged revisions 383631 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	cfd2b244f7	Corrected some module issues introduced by r383579. ... When I moved res_json.c to json.c, I left the MODULE_INFO stuff in there, which was interesting if you ran module show. I also forgot to call what was in module_load() from asterisk main(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	cf9324b25e	Move more channel events to Stasis; move res_json.c to main/json.c. ... This patch started out simply as fixing the bouncing tests introduced in r382685, but required some other changes to give it a decent implementation. To fix the bouncing tests, the UserEvent and Newexten AMI events needed to be refactored to dispatch via Stasis. Dispatching directly to AMI resulted in those events sometimes getting ahead of the associated Newchannel events, which would understandably confuse anyone. I found that instead of creating a zillion different message types and structures associated with them, it would be preferable to define a message type that has a channel snapshot and a blob of structured data with a small bit of additional information. The JSON object model provides a very nice way of representing structured data, so I went with that. * Move JSON support from res_json.c to main/json.c * Made libjansson-dev a required dependency * Added an ast_channel_blob message type, which has a channel snapshot and JSON blob of data. * Changed UserEvent and Newexten events so that they are dispatched via ast_channel_blob messages on the channel's topic. * Got rid of the ast_channel_varset message; used ast_channel_blob instead. * Extracted the manager functions converting Stasis channel events to AMI events into manager_channel.c. (issue ASTERISK-21096) Review: https://reviewboard.asterisk.org/r/2381/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	401f7c1880	Fix skinny voicemail indication issues. ... Unsubscribe from MWI stasis event on channel reload. (closes issue ASTERISK-21216) Reported by: wedhorn Tested by: snuffy, myself Patches: skinny-mwiind02.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago